Grandstream Networks GXE502X user manual Version

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GXE502X User Manual 7/9/2008

oSession Keep Alive: This field allows users to enable or disable usage of the session

timer. This is enabled by default to: Automatic/Session Timer.

oSession Expiration: The session timer enables SIP sessions to be periodically “refreshed” via a SIP request. If enabled, before the session interval expires, the GXE502X will send a SIP re-INVITE message to the SIP server, if this message is not replied the session will be terminated. This will avoid stalled sessions occupying the channel/bandwidth. Session Expiration is the timer (in seconds) at which the session is considered timed out if no successful session refresh transaction occurs beforehand. The default value is 180 seconds. Incorrect configurations could cause

calls to get dropped.

oMin-SE:The minimum session expiration (in seconds). The default value is 90 seconds.

oUse DNS SRV: Default is No. Select “Yes” if the domain name of the SIP server or Outbound Proxy is compliant with RFC2782. The SIP trunk will not function properly

if this field is not configured correctly.

oUnregister on Reboot: Default is No. If set to “Yes”, The GXE502X will first send a registration request to remove all previous or multiple bindings by adding “*” in the SIP Contact Header. Please use this feature ONLY if the proxy supports removal binding requests. Otherwise the SIP trunk will not function properly.

oRegister Active: This parameter controls whether the GXE502X needs to send REGISTER messages to the Proxy Server before making or receiving calls. The default setting is “Yes” (for most dynamic SIP trunks).

Note:

For security reasons: If this is set to “No” (for most static SIP trunks) any incoming request MUST match the SIP server IP or the FQDN resolution must match the sending IP address.

oCBCOM Encryption: If set to Yes, RTP will be encrypted as per the algorithms of CBCOM. The Default setting is No.

oDID Switch: This field is used to route incoming calls when a DID (Direct Inward Dialing #) is used instead of the account ID. It is very helpful when there are several DIDs related to the same (static) trunk provided by your SIP Trunk Service Provider.

Note:

All SIP requests are verified by the GXE502X so the User Part of the SIP URI has to match the account ID or the DIDs. For outgoing calls when DID switch is enabled, the DID number in the list will be used in the From, Contact and PAI (P-Asserted- Identity) value.

If a prepend prefix is configured, it will also be used as prefix for the DIDs.

For example, if the DID is +16175669300 and for outgoing calls the service provider requires “+”, then configure the prepend prefix to “+” and add the DID 16175669300. Then +16175669300 will be used to match both incoming calls and also outgoing calls.

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Contents Version Table of Contents Equipment Packaging Extension ConfigurationIntroduction Phone ExtensionsLocal Ippbx Number Express SetupExtension Length Extension Digit PrefixVersion Busy Lamp Field and Message Waiting Indication General SettingsExtension Number for Paging Trunk ConfigurationAuto Provisioning Extensions after Express Setup Extension Number for Internal Phone/FAX portsEnable Current Disconnect Default is Yes Internal Pstn Trunk Line Configuration via FXO portsCurrent Disconnect Threshold Version Internal Phone/Fax Port Line Call ControlSIP trunk Version Version Version External Pstn Trunk Line Version Conference Bridge Hunt/Ring Group Configuration Group Name Enter a name to identify this hunt/ring group Settings sub page of the Phone Extensions menuAuto-Attendant Configuration Voice Menu Configuration Version Playing Rules Configuration System Settings In-Queue Announcements Configuration Call Queues ConfigurationCall Queues Configuration Version Version System Configuration Networking SettingWAN Setting WAN-side Access and Security Using Dynamic DNSConfiguring port forwarding Route Configuration System Setting Setting the Administrator login passwordAdministrator Contact information Start of UDP Port for Media Transfer SIP Static Mapped IP and port ConfigurationVoicemail-to-Email Configuration SIP Static Mapped WAN Port for CBComMTZ+6MDT+5, Time Zone Self-Defined Time Zone ConfigurationManual Selection of Fax System Music MoH SelectionVoicemail/Fax mail box storage quota Version Feature Codes ƒ Call Forward Status Inquiry Version Firmware Upgrade Syslog Configuration Backup & Restore ConfigurationSystem Level Backup & Restore Peer Systems Advanced Options ConfigurationVersion Template Upload Reset & Reboot Status Version Version Reports System StatisticsCall Statistics Version Version Version Version Call Records Downloading Call Records Configuring Voicemail through the IVR Voicemail ConfigurationVersion Personal Web Portal Version